2. The computational requirement gets to grow and often by more than a factor of 2. That is why people that bought into 192KHz often ended buying very expansive accelerator cards, and still came short.

3. That is the big one: There is a tradoff between speed and accuracy. Clearly, the accuracy of a 10Hz system is great, but it is too slow for audio. The accuracy of 1GHz is much poorer, and it is too fast for audio. The question is - what is the optimum rate?

It is not true that faster is better. It is not true that more is always better. A 6 foot person weighing 100lb is too thin, but the same person weighing 500lb is too heavy. There is such a thing as OPTIMAL RATE. In the case of audio, it is all about what people can hear. That is what dictates why most mic's and speakers are optimized to about 20-20KHz, not 20-96KHz. The same factors should apply to converters.

The speed accuracy tradoff is one of the general engineering concepts, and it manifests itself many ways. Most of them are practical, such as "you can charge the cap more accurately if you have more time", or "the amplifier will settle to a more accurate value if you give it more time". But with modern converters, mostly based on sigma delta, the tradoff starts on paper, before we get to "real world" circuits. The basic given set of design parameters for a sigma delta converter are 1. oversampling ratio 2. filter order 3. number of quantizer bits.
Say you have a given set of parameters.
You can design for the best 0-24KHz audio bandwidth
You can have less precision but more bandwidth 0-48KHz
You can have even less precision but more bandwidth 0-96KHz

This was regarding the paper design stage of sigma delta. Than you get into the real world circuitry and face the same tradoffs again...

There is no escape from speed vs accuracy tradoffs, sigma delta or not...

Regards
Dan Lavry"

" I know that some of the concepts regarding sampling are NOT intuitive. It is difficult to explain that more samples are not better in a world where more pixels are better, but the fact remains, samples are not pixels and there are issues that are not easy to convey to people that did not chose to take an EE or math career. I wrote my paper to try to simplify things, but I guess it is still too difficult for many to follow.
So let’s just say that Nyquist was right, and we have 100 years of hand on experience, including test equipment, the communication industry, digital video, digital audio and much more.
And even without that experience, it is proven solidly to be mathematically correct that more samples than needed (as indicated by Nyquist) are going to add ZERO content, and are totally redundant.

Regarding that speed – accuracy tradeoff, that is easier to understand. Analogies can be misleading, but say you take on a task to color a picture with crayons and “stay within the lines”. The picture is intricate. I bet doing it in 10 seconds will be a lot less accurate than if you took 10 minutes. The same statement applies towards so many things. Devices and circuits also have speed limitations (and speed is in fact bandwidth). A given size capacitor takes time to charge, a logic gate takes time to change states and so on. Doing things fast goes against doing things accurately. Devices and circuits can be optimized for maximum speed, power, accuracy and more. They are most often optimized to provide a combination of acceptable tradeoff. When you relax on one requirement, you end up with more “breathing room” for other requirements.

Regarding the sigma delta design, yes, in theory you give up accuracy for speed. The noise shaping concept is about moving noise from a frequency range you wish to use for the signal, to other frequencies. Think of it as digging a hole. You can either dig a deep hole of small diameter, or very shallow hole of a large diameter. It is the same amount of dirt, but a different result. The depth of the hole is analogues to the accuracy, the diameter represent the bandwidth. Do you want great 20KHz or not so great 100KHz?

That answers your question about paper design. But I am an engineer and therefore equally interested in the real parts and circuits. Speed vs accuracy is a solid concept. Speed vs power is another one and there are others. Those concepts are no different than the first law of thermodynamics – never proven but no one so far came up with a single example to contradict it.

For me 48 is not worse than 96 in 99,99% of applications, but it might strongly depend on requirements I put in front of system. At the end of the day, same project (let say 24 tracks) put through same outboards with only difference in sampling rate (same converters naturally) sounds EQUALLY to me.
ONLY ONE DIFFERENCE.
If I make more than usual ADC-DAC-ADC (triple conversion) for any reason on various or all tracks, 96 kHz MAY slightly improve high end transparency, but I NEVER do it, just do not need it.
Even comparison 48 to 192 DOES NOT bring essential improvement in favour of 192. It is always more like self-suggestion than something we can hear.
Maybe above will be different on 50 k+ main monitors and highest end rooms, but I do not know many people who really need it for real life work (exception are recordings of music that probably need more DSD than PCM in any available format today).

44 or 48k is normally just fine - I'll use 88.1 or 96 only if there's going to be a number of plugins on the signals, since some plugins have better implementations at higher sampling rates. Or, with certain prosumer units like a DIGI002 I might use higher sampling rates since the 44.1 and 48k implementations produce audible problems in the high frequencies. But that's not an issue of higher sampling rates being better - it's one of particular software or ADC implementation.

Originally posted by ISedlacek I found that with a high quality AD (I have Lavry Blue), 44k sounds a way more pleasant (more round and "analogue") than 96 k . So I fully returned to 44 k. Simply perfect.

These are my thoughts exactly.

It's not that 192khz sounds better, its the overall design of the converters that sound better. Given that Digital audio is only 20 something years old and that 24bit, 48khz is only 15 or so years old it is not surprising that they are still creating better and better sounding converters at 24bit, 48khz sample rates. It is still an emerging and evolving technology.

Do you think a 888 would have sounded better if it had the capabilities of 192khz audio back when it was released? I doubt it, we would still be looking back like we are today saying it sounded terrible. I don't think the sample rate has anything to do with it... its all in the design.

Originally posted by ISedlacek I found that with a high quality AD (I have Lavry Blue), 44k sounds a way more pleasant (more round and "analogue") than 96 k . So I fully returned to 44 k. Simply perfect.

There's more aliasing errors when dealing with the conversion itself. As far as sound quality in digital processing, higher sampling rates are still better theroretically because there's no caps being charged or opamps involved. I think I remember hearding that when CD's were developed, sony believed that 60 kHz or so was optimal but digital processing was not powerful enough.

Originally posted by Revelation Think of it as digging a hole. You can either dig a deep hole of small diameter, or very shallow hole of a large diameter. It is the same amount of dirt, but a different result.

depends if you are digging for a well or planting a bush. that analogy is bunk.

i must also acknowledge taht every album i have heard lately recorded digitally and thought 'WOW' was done at 96k.... after the fact of hearing it and the resulting reaction.

Here's a ex BBC engineers opinion.. I tend to agree with Hugh 100% of the time. He's just too smart for me.

"Question"
Why do I here such a massive diference in the high frequencies when I record drums at 88.2K over 44.1K. I've even tried recording filtering off everthing over 18K to see if it still sounded better and it did.

If the waveforms are being accurately reconstructed why do I hear such a drastic difference.

To re-quote the post above: because of cheap parts! You don't say what converters (and other equipment) you are using, but I'd wager we are not talking Prism or dCS. Essentially what you are hearing are the audible artefacts of duff converters !

The simple fact is that the vast majority of converters don't do what the theory calls on them to do when operating at 44.1 or 48kHz. The anti-alias/reconstrution filters have an audible impact onthe pass band when they shouldn't. They introduce amplitude ripples and horrendous phase distortions. And if they are designed to minimise these aspects, the transition band is insufficiently steep and the stop band insufficiently attenuated, resulting in aliasing distortions.

They don't work properly at 88.2 or 96kHz either, but ther is so little audio signal energy close to the turnover points that the problems don't manifest.

However, technology continues to move on, and a recent AES paper by M Craven has proposed the application of a technique used in radio astronomy to correct for some of the inherent deficiencies of the digital filter designs used in anti-alias and reconstruction filters. Some manufacturers are already starting to use this new idea, and the results look (and sound) impressive.

Going back to the thought at the start of the thread, there are distinct and audible advantages to operating at 88.2 or 96kHz with current, available converter technologies compared with the results obtained with run of the mill 44.1/48kHz converters. I am very sceptical about any advantages of working at rates higher than 96, but remain open to persuasion -- I know and respect several engineers who claim additional sonic benefits.

Some of the really high end converters (the likes of Prism and dCS to name but two UK ones) can achieve the same (or better) sonic quality as budget and mid-price converters operating at 96kHz -- suggesting that when engineered properly, the Nyquist theory really does hold up.

As always, inthe affordable end of the market, quality is limited by cheap parts, cut corners, and poor design implementations. Nothing new in that -- the same problems affect analogue products too. Why else would a Neve console sound so much better than a Mackie or a Soundcraft?

It is also true that some of the these ultrasonic components can interact in the air to produce audible intermodulation products. I think this is one reason why miking a string section from a reasonable distance always produces a richer, more pleasing sound that close miking each instrument and mixing in a desk!

Also, very few microphones have a response that extends significantly above 25kHz or so, and the same for loudspeakers. So in most cases, the mic is not capturing ultrasonic energy even if it is there, and neither can the speaker reproduce it.

But, the high end roll off in both cases is gentle -- 6dB octave, typically -- which means there is little phase distortion.

Contrast that with digital converters with brickwall rol-offs operating at 44.1 or 48kHz sample rates. These inherently cause horrendous amounts of phase distortion around the turnover freuqnecy, and that, I think, it what our ears pick up on as 'the digital sound' that many don't like. Move the sampling rate up to 96kHz, and while the phase distortion still happens, it is now way outside the hearing range and so the sound appears to have improved.

But it's not because higher harmonics are being captured, or conveyed, or reproduced. It's because one of the unpleasant artefacts of digital encoding has been circumvented.

Lavry Blue and Mytek are similar in quality with their own sound. Which one is better is personal preference. The Lavry Gold is on the same level as Prisims. Very high end, but for many of us users on these forums, it's out of our reach

Originally posted by JTR So you're liking the Lavry Blue more than the Mytek you had before?

What about the Lavry Gold?
Were you able to compare with the Blue?
Any comments?

Yes, I like Lavry sound much more. I would not say it is a question of "different flavour", it is simply a different sound class. I never heard Lavry Gold. And to say the truth, I have even no desire to hear it .. Even if it sounds yet slightly better than Blue (but who knows ?) , I am not going to buy it. I consider such price levels ($7000 for 2 AD or DA channels) as quite insane ... How much will it be worth in few years ? $ 700 or $ 70 ? This is not a preamp or microphone.

Sounds like there may be a market for an ADC with a 96k initial sample conversion with an FIR to downconvert to 48k or 44.1k for output.

I still think that, if doing any processing, keeping everything at 96k until the two (or 5.1) track output pays some dividends. If you are doing a straight-to-master recording of an ensemble, then recording at 48k should be fine (however you can best get there before saving the bits).

I specifically AB compared samples of various instruments and vocals recorded in 44 and 48 k (48k samples were then converted to 44 with Voxengo r8brain). I could not hear ANY sound difference worth mentioning. Really nothing which would make even a slightest reason to record in 48 instead of 44 ... (in case of making a music CD). Recording chain: Schoeps (Neumann) - Millennia - Lavry AD..Listening chain -Lavry DA - B&W Matrix S3 801

Earlier in this thread someone implied that Lavry was discounting higher sample rates because his converters cannot support those rates.

Quite the opposite is true. He has chosen to design converters that agree with his philosophy rather than make what he can sell the most of, which is either a brave and principled stand, or short-sighted marketing-- depending on your view! In fact, it is hard to find chips that are NOT 192. He is doing something very rare today-- putting your money where your mouth is!

FWIW I experimented with 176.2 v 88.2 about a year ago with very complex "classical" material.

I discovered that:

1) more than 6 channels brought my DAW to its knees from throughput issues (and it is a 2.8GHz P4)

2) 60MB/channel per minute REALLY eats up real estate

3) there was no audible reason to go past 88.2 (never mind the final bounce to redbook) Maybe I need an ears upgrade? (Monitors are B&W matrix 801 III)

Granted, the lack of sonic side effects at 88.2 rather than the brickwall 44.1 IS audible, but as the point was made above, good filter design vs cheapo is huge. This also ties in with the fact that a Prism, Meitner, dCS or Lavry can sound better at 44.1 than many lesser ones at any other rate.

I finally realized that even if I was honest and admitted that my quest for 192 was marketing-driven, the fact was that 99% of all clients don't know or care, and cannot hear the difference. The only benefit was to hard drive manufacturers!

From what I've heard from you guys, and read in Bob Katz's book on Mastering audio, if you've got a good converter , with well designed low pass filters, you can get by with 48 khz sample rates for your recordings, and they will sound just as good as if you recorded at 96khz. The only thing is this.... I think most mastering houses , as stated by a preference of Bob Katz, would prefer to receive a 96 khz file because in the process of digital processing, the higher qualtiy can be maintained. I personally am planning on using 96 khz, since I feel that my plugins and digital processing will not degrade the signal as bad as if I recorded at 48 khz.... at least ..this is my theory.... (hunch) if you will, and also, I like to be able to tell customers that I can record them and deliver a DVD-A if they wish. Another idea... which I'm not too sure about the credibility of this.... will mixing in the digital domain benefit from using 96khz ? The new version of Samplitude is shipping now, which boasts on their website that their mixing algorhythms can compete with high end consoles. I'm not sure if there's any truth to this, but if it is true, then 96 khz would probably yield better mixes if your mixing digitally....
My two cents.... but take it with a grain of salt since I'm just a beginner !!

it´s based on a vintage theory by Harold Nyquist, 1928! So, if i got it right, in theory, 44,1kHz is enough. In practice it might be better to use higher than theoretical sampling rates. And, in the real world, it depends on which practice.

In theory, you don't need a sampling rate more than twice the greatest frequency of interest. But you have to have a perfect "brick-wall" filter to capture a signal at 1/2 the sample rate.

In practice, you can't make a perfect "brick-wall" filter where the pass-band is totally unaffected and the reject-band has infinite attenuation. In reality, there is a transition-band in analog filters and the pass-band does get affected by high-order analog filters.

There is always some phase effect near the transition-band of analog filters, but the biggest trade-off for ADCs is going to be between attenuation of the higher frequencies and ringing of the filter.

With higher sampling rates, you can move the filter to a higher frequency where these effects can't be heard.